tag:blogger.com,1999:blog-80090545067565429172018-08-09T02:03:39.540-04:00Adventures in VoIPAll the information I am learning as I attempt to navigate the exciting world of telephonyAaron Dycknoreply@blogger.comBlogger53125tag:blogger.com,1999:blog-8009054506756542917.post-51570874684387545412018-05-28T18:21:00.001-04:002018-05-28T18:21:31.421-04:00Upgrading the IP OfficeI'm not going to tell you how to upgrade your IP Office, that's what the official training is for. I am, however, happy to discuss the various issues that I have experienced while performing upgrades, and the steps I take to ensure that the upgrade goes smoothly.<br /><br /><h4>File Write Errors</h4><div>I often see people struggling when transferring files to the IP Office during an upgrade. More times than I'd like to count I've seen the system pop up with an error stating "405 Method Not Allowed".&nbsp; While there are a number of things that can cause this issue the most common, by far, is the File Writer IP field in the system tab. If you're getting this error just check there and see if there is an address populated. I usually set the address to 0.0.0.0, which allows the writing of files from any address. While this may present a security risk, as long as you follow strong password management and don't leave any of the default passwords in place the risk should be next to zero. Alternatively you could specify the address of the PC being used to perform the upgrade, restricting the ability to write to the memory card to just that device. The third option is to set the address to 255.255.255.255, which will cause the system to update the address in this field the next time embedded file management is used to reflect the address of the system that is using embedded file management. </div><div><br /></div><div>In some cases, even after making the change to the File Writer IP address, some installations still don't proceed. In these cases I've seen a couple of solutions. In some instances the system will allow the completion of the upgrade to the core equipment, and not the update of the files on the SD Card. When this happens it is often possible to simply log in to Embedded File Management and upload the system files from there. If that doesn't work, however, the cause is usually a corrupted SD Card. This happens most often on systems that have been through a number of upgrades, or potentially if you've forgotten that you have to step up from any release below 8.1(65) to 8.1(65) or 9.0 prior to upgrading to the current release. When this happens the best plan is to pop out the SD Card, put it in your laptop, and recreate the SD Card. Once this is complete pop it back in to the system and push your backed up config (and any embedded voicemail files, if applicable) back to the system. Once this is complete you should be able to access the system just fine.</div><div><br /></div><h4>Server Edition and Application Server Upgrades</h4><div>Server Edition and Application Server upgrades appear to take forever, frequently appearing to stop around the 92% complete mark. This has caused panic for so many of my colleagues over the years. The first time I encountered this I panicked and rebooted the server. Let me tell you, this is a mistake! Now I've learned a thing or two and I have found a better way to keep an eye on the progress than simply looking at that 92% for what seems like an eternity. Log in to the server as root and type the following command:</div><blockquote class="tr_bq"><span style="font-family: Courier New, Courier, monospace;">tail -f /var/log/yum.log</span></blockquote><div>This command will continuously scroll the output of the YUM application. YUM, or the Yellowdog Updater, Modified, is the Linux program that the IP Office uses to manage the installation of software. The yum.log file will scroll frequently even if the Web Manager progress indicator hasn't moved. This should keep your mind at ease while you wait.</div><div><br /></div><div>It is possible that the updater may fail. Avaya has documented issues with the upgrade failing, and has even written code into the update software to identify issues with YUM failing to complete the upgrade. If you see the error "yum process died before completing its job" you have experienced an issue with the /boot/ partition being full. This happens, again, on systems that have been upgraded multiple times. If your system has been upgraded to at least 9.1.7 you should not experience this issue. If you do have this issue after 9.1.7 engage Avaya support right away, and anticipate a rebuild. If you are at a release older than 9.1.7 there is a quick fix. Download the file UpgradeKernelFix_v2.zip from Avaya. This zip file contains a patch that will allow you to expand the /boot/ partition. Extract&nbsp;UpgradeKernelFix_v2.sh from the zip file and upload it to your server using WinSCP, or your favorite SFTP client. Browse to the location of the file as root and type chmod +x&nbsp;UpgradeKernelFix_v2.sh - this makes the file executable. Run the file with ./UpgradeKernelFix_v2.sh and you should be off to the races. Of course, you will have to start the upgrade again. This issue should only affect systems running on VMWare, however I wouldn't rule it out for bare metal servers.</div><div><br /></div><div><br /></div><div><br /></div><div><br /></div><div>I can not stress enough the importance of making a full backup before you begin. For an application server this means assessing the applications used and ensuring that all application data is backed up off the system before you begin. For systems using embedded voicemail this means ensuring that you save all the dynamic contents of the SD Card before you even think of starting.&nbsp;</div><div><br /></div><div>Let me know if you've experienced any other errors, and if so, what is your resolution. I'll try to help out anyone who is stuck if I can!</div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com1tag:blogger.com,1999:blog-8009054506756542917.post-47507055135570091652017-10-25T13:37:00.000-04:002017-10-25T13:37:22.707-04:00SIP StandardThe VoIP world is dominated by SIP lately. SIP is a standard that is defined by a number of RFCs but searching through here for answers to questions is ridiculously difficult. I would like to use this post to answer any SIP questions that I come across in my daily travels. If you have a question that has not been answered feel free to drop it in the comments. I'll do my very best to find an answer for you!<br /><br />Q: What is the maximum size of a SIP packet?<br />A: The maximum allowed size is equal to the maximum size of a UDP packet, or 65,507 bytes (65,535B - 20 B IP Header -8 B UDP Header). Some software may not accept packets of this size. Often software will limit the packet size to 4096B.<br /><br />Q: What is a SIP User Agent (UA)?<br />A: A User Agent refers to a component in the SIP communication between two devices. There are two types of User Agents: Client and Server. A UA Client (UAC) sends a SIP request and a UA Server (UAS) sends a response. Most of the time a SIP server or endpoint will act as both a UAC and a UAS, depending on the circumstance. A call may terminate to a phone in which case the SIP server is the UAC and the phone is the UAS. The phone may then place the call on hold or conference another user, at which point the phone sends the request and becomes the UAC.<br /><br />Q: What is a B2BUA?<br />A: A B2BUA (Back to Back User Agent) is a device that receives a request from a UAC and forwards that request out to another device, acting initially as the UAS and as the second leg as a UAC. A B2BUA will remain in the middle of the conversation so that the endpoints do not communicate directly for signaling purposes. It is still possible to have a B2BUA with direct media path for RTP packets.<br /><br />Q: What is the maximum speed for faxing with SIP?<br />A: Contrary to what most people believe, there is no difference in the maximum speed of a fax machine on analog or PRI versus SIP. The SIP standard doesn't care - this is a function of the codec negotiation. But that doesn't answer the real question here. When I'm setting up a PBX I always suggest the customer limit their fax speed to 14,400bps. The PBX will see the 14,400 speed (or lower) and will use the T.38 codec for faxing. Most phone systems will only allow T.38 if the PBX will be holding up the RTP (i.e. no direct media path) as the system will switch to T.38 when it detects the fax tones being sent from the originating fax machine. Modern fax machines (also known as Super G3 fax machines) support speeds of up to 38,400bps, although the realized throughput is often lower. When using a Super G3 fax machine the codec will negotiate to G.711. Network reliability is incredibly important when using Super G3 speeds. These machines are incredibly intolerant to jitter and packet loss.<br /><br /><br />Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com1tag:blogger.com,1999:blog-8009054506756542917.post-1833947093583949682017-08-22T11:29:00.000-04:002017-08-22T11:29:45.505-04:00IP Office DHCP ScopeThe IP Office has a built-in DHCP server that can be configured to provide addresses for all devices or just for Avaya phones, however many companies have their own DHCP servers that they manage. The IP Office phones look for one of two DHCP Options for their information, either 176 or 242. Both of these options use similar formats.<br /><br />Option 176 is used for legacy IP Office phones (4600 and 5600 series phones). Option 242 is used for 1600 and 9600 series phones. The DHCP option provides the IP Office address, the H.323 port number, the TFTP or HTTP server used, and (optionally) VLAN information.<br /><br />The options look like this:<br /><br /><blockquote class="tr_bq"><span style="font-family: Courier New, Courier, monospace; font-size: x-small;">OPTION 176 &nbsp;MCIPADD=xxx.xxx.xxx.xxx,MCIPORT=yyyy,TFTPSERVER=zzz.zzz.zzz.zzz</span><span style="font-family: Courier New, Courier, monospace; font-size: x-small;">OPTION 242 &nbsp;MCIPADD=xxx.xxx.xxx.xxx,MCIPORT=yyyy,HTTPSERVER=zzz.zzz.zzz.zzz</span></blockquote><br />xxx.xxx.xxx.xxx is the IP Address of your IP Office that the phones will use for registration, and yyyy is the port number (almost always 1719). The TFTP or HTTP server address is almost always the same as the IP Office address, however some deployments may use an external server to support additional simultaneous connections.<br /><br />In a scenario where you have multiple servers in a resilient environment you can have multiple entries for MCIPADD - simply separate them with a comma:<br /><blockquote class="tr_bq"><span style="font-family: Courier New, Courier, monospace; font-size: x-small;">OPTION 242 MCIPADD=192.168.42.1,192.168.42.2,MCIPORT=1719,HTTPSERVER=192.168.42.1,192.168.42.2</span></blockquote>With Option 242 you can specify a folder on the HTTP server that contains the files needed by the IP Phone using the HTTPDIR=path_to/files.<br /><br />Keep in mind that there is a limit of 127 characters in a DHCP Offer. If the scope would exceed this you can use OPTION 66 to specify the TFTP Server for Option 176.<br /><br />When using VLANs to separate voice and data traffic you need to ensure that the VLAN details are specified in the default VLAN:<br />L2Q=1 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;enables VLAN tagging.<br />L2QVLAN=xxx &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; where xxx is the VLAN ID for your voice VLAN.<br />In this scenario you would have your options as follows:<br />Default DHCP Offer:<br /><blockquote class="tr_bq"><span style="font-family: Courier New, Courier, monospace; font-size: x-small;">OPTION 242 L2Q=1,L2QVLAN=200</span></blockquote>Voice VLAN DHCP Offer:<br /><blockquote class="tr_bq"><span style="font-family: Courier New, Courier, monospace; font-size: x-small;">OPTION 242 MCIPADD=192.168.42.1,MCIPORT=1719,HTTPSERVER=192.168.42.1</span></blockquote>The IP Phones will pick up their VLAN from the default offer then request a new DHCP scope in the correct VLAN, at which point they will receive their configuration from the voice VLAN.<br /><br /><br />Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-39804349670446410632017-08-16T13:09:00.000-04:002017-08-16T13:09:58.681-04:00IP Office Personal Directory with pausesToday I had someone call me to say they were trying to create a personal directory entry in their IP Office for someone with an external number and and extension. I couldn't conceive of why there would be any problem with this until I learned that the Personal Directory only allows entry of keypad keys (0-9, *, #). Since none of the allowable entries will make the call pause I had to find a way to make this work.<br /><br />After some thought about how I could make this work for my customer I came up with a plan. I would try to dial a short code that would make the call for me!<br /><br />First I created the PD entry:<br /><br /><div class="separator" style="clear: both; text-align: center;"><a href="https://3.bp.blogspot.com/--jmRMuFvFTM/WZR5oXQJNmI/AAAAAAAAcv8/33GryB6Uw8IYhCGlAd3f_WjxzwY-l2MEQCLcBGAs/s1600/PD%2BEntry.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="120" data-original-width="439" height="87" src="https://3.bp.blogspot.com/--jmRMuFvFTM/WZR5oXQJNmI/AAAAAAAAcv8/33GryB6Uw8IYhCGlAd3f_WjxzwY-l2MEQCLcBGAs/s320/PD%2BEntry.png" width="320" /></a></div><div class="separator" style="clear: both; text-align: center;"><br /></div>Then I create a corresponding Dial Direct short code with the number I have put into the PD:<br /><br /><div class="separator" style="clear: both; text-align: center;"><a href="https://4.bp.blogspot.com/-QInzSmmC94g/WZR6KpjJcGI/AAAAAAAAcwE/QdKFNe7LEcQXyt7x_QBclQlBzPgs8yOkACLcBGAs/s1600/PD%2BShort%2BCode.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" data-original-height="292" data-original-width="790" height="118" src="https://4.bp.blogspot.com/-QInzSmmC94g/WZR6KpjJcGI/AAAAAAAAcwE/QdKFNe7LEcQXyt7x_QBclQlBzPgs8yOkACLcBGAs/s320/PD%2BShort%2BCode.png" width="320" /></a></div><div class="separator" style="clear: both; text-align: center;"><br /></div>Once I had followed these quick and easy steps I was able to make a call from my personal directory to an extension. I was surprised that Avaya hadn't allowed pauses in the directory but the IP Office always provides a way! &nbsp;It is not necessary to use the phone number, I simply used this method to ensure that I was able to track what numbers were being manipulated at a glance.<br />Make sure to include the dial prefix if it is expected in the ARS. If it is not needed in the ARS you don't need to include it in the short code.<br /><br /><br />Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-16362176718898099932015-04-17T11:41:00.002-04:002015-04-17T11:55:29.962-04:00Network failures and the Dell R620So we recently installed a Dell R620 server with IP Office Server Edition 9.1. While troubleshooting an unrelated issue we came up with a big problem: The device could no longer be reached on the network! Well that's a huge problem since we have IP phones and all the bells and whistles on this server. Avaya support wasn't able to resolve the issue because...guess what? They couldn't get into it remotely...<br /><br />So we replaced the server and I took a look once we had it back in the lab. It turns out that the devices had been renamed. ETH0 switched with ETH2 and ETH1 switched with ETH3. By plugging in to Port 3 (which should have been ETH2) and enabling the port (#ip link set up) I was able to get connectivity right away. But how could this have happened, and what can I do to make it work with the correct network port? It turns out there is an issue with the way CentOS uses UDEV to create network interfaces. So how do you fix it? Well it's not too hard. I was able to make it work exactly as it did before by looking at a few files and changing one.<br /><br />In the root folder there are four interesting files. They are ifcfg-eth0, ifcfg-eth1, ifcfg-eth2, and ifcfg-eth3. These files each contain a single line of text with the MAC address of the server:<br /><div class="MsoNoSpacing" style="margin-left: 1.0in;"><span style="font-family: &quot;Courier New&quot;; font-size: 10.0pt;"><br /></span></div><div class="MsoNoSpacing" style="margin-left: 1.0in;"><span style="font-family: &quot;Courier New&quot;; font-size: 10.0pt;">HWADDR="b8:2a:72:dc:17:28"<o:p></o:p></span></div><div class="MsoNoSpacing" style="margin-left: 1.0in;"><span style="font-family: &quot;Courier New&quot;; font-size: 10.0pt;"><br /></span></div>Trust these files. They are the gospel. I was also able to find the MAC address of eth0 (Port 1) printed on the circuit board inside the server. The four addresses should be sequential, but keep in mind that they are sequential in HEXADECIMAL. So...29 does not go to 30, 29 goes to 2a.<br /><br />So you may wonder, now that we have these MAC addresses, what are we going to do with them?<br /><br />That's easy. Use your favourite text editor (I personally like NANO) to edit the file /etc/udev/rules.d/70-persistent-net.rules. You'll see your interfaces and, I bet, incorrect MAC addresses. Or at least MAC addresses that are not associated with the correct device name. Simply change the MAC addresses to the correct address for each device and you're on your way.<br /><br />For example, change:<br /><blockquote class="tr_bq"><blockquote class="tr_bq" style="margin-bottom: .0001pt; margin-bottom: 0in; margin-left: .25in; margin-right: .25in; margin-top: 0in;"><span style="font-family: &quot;Courier New&quot;; font-size: 10.0pt;">SUBSYSTEM=="net", ACTION=="add", DRIVERS=="?*", ATTR{address}=="b8:2a:72:dc:17:2a", ATTR{type}=="1", KERNEL=="eth*", NAME="eth0"</span></blockquote></blockquote>&nbsp;to:<br /><blockquote class="tr_bq"></blockquote><br /><blockquote class="tr_bq" style="margin-bottom: .0001pt; margin-bottom: 0in; margin-left: .25in; margin-right: .25in; margin-top: 0in;"><span style="font-family: &quot;Courier New&quot;; font-size: 10.0pt;">SUBSYSTEM=="net", ACTION=="add", DRIVERS=="?*", ATTR{address}=="b8:2a:72:dc:17:<b>28</b>", ATTR{type}=="1", KERNEL=="eth*", NAME="eth0"</span></blockquote><br />Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com2tag:blogger.com,1999:blog-8009054506756542917.post-80907563713957746252015-01-09T10:29:00.000-05:002015-01-09T10:29:12.112-05:00Security Certificates - 9.1<div class="MsoNormal"><span style="font-family: inherit;">With the release of IP Office 9.1 there have been enhancements made to the way security is handled. When deploying an IP Office Server Edition or Select Server Edition for a customer it is best practice to have them provide a fully qualified domain name or a machine name to use for the security certificate. The IP Office can be configured with a valid host name and the certificate can be imported into the Trusted Root Certification Authority certificate store. When accessing the system by the proper host name with the certificate properly stored there will be no security warnings while accessing the page.<o:p></o:p></span></div><div class="MsoNormal"><span style="font-family: inherit;"><br /></span></div><div class="MsoNormal"><span style="font-family: inherit;">When defining the hostname for the IP Office you need to either enter the FQDN that will be used to access the system or the IP address that will be used to access the system. In the case of my example I used the IP address of 192.168.11.11 as the host name. If you are using a fully qualified domain name (FQDN) or a server name (NetBIOS) you will want to make sure it resolves with your DNS server or you will see a certificate mismatch error.<o:p></o:p></span></div><div class="separator" style="clear: both; text-align: center;"><a href="http://2.bp.blogspot.com/-1Ks0QbgffRM/VK_x8UKXAII/AAAAAAAAQYU/R1rqgcpQo8k/s1600/IPO-Cert-1.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><span style="font-family: inherit;"><img border="0" src="http://2.bp.blogspot.com/-1Ks0QbgffRM/VK_x8UKXAII/AAAAAAAAQYU/R1rqgcpQo8k/s1600/IPO-Cert-1.png" height="256" width="320" /></span></a></div><div class="MsoNormal"><span style="font-family: inherit;"><br /></span></div><div class="MsoNormal"><span style="font-family: inherit;">To use a self-signed certificate we will select “Generate New”:<o:p></o:p></span></div><div class="MsoNormal"><span style="font-family: inherit;"><br /></span></div><div class="separator" style="clear: both; text-align: center;"><span style="font-family: inherit;"><a href="http://2.bp.blogspot.com/-xuPg_fTt_1Y/VK_yFTA1KHI/AAAAAAAAQYc/s3GDgfan4xA/s1600/IPO-Cert-2.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="http://2.bp.blogspot.com/-xuPg_fTt_1Y/VK_yFTA1KHI/AAAAAAAAQYc/s3GDgfan4xA/s1600/IPO-Cert-2.png" height="256" width="320" /></a><br /></span></div><div class="MsoNormal"><span style="font-family: inherit;">After you click Next you will see the following warning:<o:p></o:p></span></div><div class="separator" style="clear: both; text-align: center;"><a href="http://4.bp.blogspot.com/-IdmEKUl0_jQ/VK_yM1Y1b8I/AAAAAAAAQYk/pIh_AIqZUiE/s1600/IPO-Cert-4.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><span style="font-family: inherit;"><img border="0" src="http://4.bp.blogspot.com/-IdmEKUl0_jQ/VK_yM1Y1b8I/AAAAAAAAQYk/pIh_AIqZUiE/s1600/IPO-Cert-4.png" height="160" width="320" /></span></a></div><div class="MsoNormal"><span style="line-height: 115%;"><span style="font-family: inherit;"><br /></span></span></div><div class="MsoNormal"><span style="line-height: 115%;"><span style="font-family: inherit;">The certificate will now be generated.&nbsp;</span></span></div><div class="MsoNormal"><span style="line-height: 115%;"><span style="font-family: inherit;"><br /></span></span></div><div class="MsoNormal"><span style="font-family: inherit;">Once the certificate has been created it is available for download. For a Windows Certificate Store you need to download the DER-Encoded certificate:<o:p></o:p></span></div><div class="MsoNormal"><span style="font-family: inherit;"><br /></span></div><div class="separator" style="clear: both; text-align: center;"><a href="http://4.bp.blogspot.com/-ju_lQ71mLe0/VK_yaqBhECI/AAAAAAAAQYs/32Hyf9oA9uU/s1600/IPO-Cert-3.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><span style="font-family: inherit;"><img border="0" src="http://4.bp.blogspot.com/-ju_lQ71mLe0/VK_yaqBhECI/AAAAAAAAQYs/32Hyf9oA9uU/s1600/IPO-Cert-3.png" height="254" width="320" /></span></a></div><div class="MsoNormal"><span style="font-family: inherit;"><br /></span></div><div class="MsoNormal"><span style="font-family: inherit;">Once you have downloaded the certificate click Apply. The process will take several minutes, after which you will be logged out of the system.&nbsp; Be sure to add the certificate you downloaded to your Trusted Root Certification Authority. If you're working with a domain this can be pushed to client systems using a group policy, or it can be added to machines individually using the Microsoft Management Console.</span><o:p></o:p></div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com1tag:blogger.com,1999:blog-8009054506756542917.post-36513257069026953642014-12-31T10:59:00.000-05:002014-12-31T10:59:04.137-05:00IP Office Registry HacksIt has come to my attention that Avaya often hides functionality in a registry entry. I'm going to keep a list here. As I learn more I will add them. Feel free to comment any that you have tucked away under your cap!<br /><br /><b>Maximum UMS Users (166 by default):</b><br />Under HKEY_LOCAL_MACHINE/SYSTEM/CurentControlSet/Services/MSExchangeIS/ParametersSystem, add a new key MaxObjsPerMapiSesion. Under the new key, create a new DWORD Value objtMesageView, and set the value to three times the required users. For example, to support 500 users, set the value to 1500.<br /><br /><b>SIP Line Template:</b><br />In IP Office 9.1 the option for SIP Line Templates was included out of the box. In 7.0 and up the option is still there, but it's hidden. There is a two-part step to enable this:<br />Navigate to File --&gt; Preferences on the IP Office Manager and select the Visual Preferences tab. Check the Enable Template Options box.<br />Under HKEY_CURRENT_USER/Software/Avaya/IP400/Manager and add a DWORD value TemplateProvisioning and set its value to 1. Reboot the server hosting the IP Office Manager.<br />You can now generate a SIP Trunk templateAaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-26728120840969513952014-12-29T15:01:00.001-05:002014-12-29T15:01:21.380-05:00No Caller ID Alarms on the IP OfficeSo the IP Office screams every time a call comes in with no caller ID received. Avaya finally decided to put in a workaround, and it's super quick and easy to implement!<br /><br />The workaround is only available in 9.1 (and up, I suppose).<br /><br />Go to the NoUser user and click on the Source Numbers tab.<br /><br />Add the source number SUPPRESS_ALARM=1<br /><br />Merge your changes and that's it! The NoCallerID Notification will be suppressed for System Monitor, System Status, Sys Log, SNMP Traps, and e-mail notifications.Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-9758201721785933732014-04-24T09:45:00.000-04:002016-06-30T13:22:54.039-04:00US Robotics Modem Configuration for CS1000So my favourite way to connect to a CS1000 for maintenance programming is with a modem. This quick and easy method requires very little effort to connect. Unfortunately a modem will not work with the CS1000 out of the box. There is a bit of programming that needs to be done.<br /><br />A standard external US Robotics modem comes with a power supply and a line cord. A DB25 cable will be required. For an Option 11 system a null modem will also be needed. It's always a good idea to head to the phone room with a handful of cables and adapters. Check to make sure the TTY is enabled and working. I usually hook up with my laptop to the TTY first to check connectivity and then proceed with hooking up the modem afterwards.<br /><br />Programming the modem is straightforward and only takes a couple of minutes. Unbox the modem and check the dip switches first. You need to set it so that 1, 3, 7, and 8 are down and the rest are up. Connect the modem to your PC using a serial cable and power it on. Using your favourite terminal program (I usually use Procomm for this but Putty would also work) set to 9600/N/8/1 check to see that the modem is responding. Enter AT and press enter, the modem should respond with OK. If you don't see OK try throwing a null modem adapter in the mix and test again. Once you have a response from the modem you can enter the configuration command:<br /><br /><div style="text-align: center;">AT&amp;B1&amp;N6&amp;W&amp;W1</div><div><br /></div><div>The modem should once again respond with OK. Once this is done power down the modem and change the DIP switch settings again. At this point you need to have 1 and 4 down, the rest up. Plug the modem into the TTY port and test. The modem should answer. If you don't see and output on the TTY then you may need to either add or remove a null modem to get everything working.&nbsp;</div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-58604900426694263362014-02-12T11:19:00.003-05:002014-02-12T11:24:52.635-05:00Upgrade: File Access ErrorI am in the middle of programming up an IP Office and I came across an issue with the phones. I plugged a phone in and I received the following message:<br /><br />Upgrade: File Access Error<br /><br />I did some digging and found the solution. I decided to share it here.<br /><br />This issue comes from the fact that the phone is not able to get the firmware from the system or wherever it is configured to go for the update. In my case I had configured the IP Office to retrieve its firmware from Manager, however I had the wrong IP Address in Manager. Basically what happened is the phone went looking for the firmware and couldn't retrieve it from the identified location. The solution is quite easy. In my case I just changed the Phone File Server Type to Memory Card. The other options would be to configure the Manager PC IP Address to reflect the location of the correct version of Manager (Making sure the IP Office can reach that PC), or to change the type to Custom and specify the IP Address of the server that contains your BIN files.<br /><br />Hopefully this helps!Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com2tag:blogger.com,1999:blog-8009054506756542917.post-61799145269542658842014-01-15T14:14:00.003-05:002017-04-10T09:15:10.505-04:00IP Office BCM CS1000 Integration with NRSI recently had to network some BCMs, and IP Office 8.1, and a CS1000 together for four-digit dialing. I figure I will share some of my knowledge here. With the release of IP Office 9.0 a Session Manager is required which greatly simplifies the implementation, as long as you know how to route calls using Session Manager.<br /><br />For the purpose of this article I'm going to assume that everyone knows the basics of how to build a SIP trunk on all three platforms. I will also assume that the appropriate licenses have been purchased and that the BCM is SIP capable.<br /><br />Let's start with the CS1000. The NRS needs to be configured You will need to create endpoints for the BCMs and IP Offices, as well as any CS1000 locations that you have. The IP Office and BCM endpoints must be configured as static SIP endpoints.Routing entries need to be created for each endpoint to correspond with your dialing plan. It may seem obvious, but you need to build SIP trunks and routes from the CS1000. <b><i>It is very important to note that the IP Office must be configured in Proxy Mode. The IP Office does not support Redirect Mode</i></b>. Once you have finished with this you are done with the easy part.<br /><br />The BCM configuration is also quite easy. Build your SIP trunks on the BCM. Configure your SIP domain and enable RTP keepalives. Calls will be routed on the Private network, so navigate to the Private tab under SIP Trunking. Ensure that your URI map matches the URI Map on your node:<br /><br /><div class="separator" style="clear: both; text-align: center;"><a href="http://2.bp.blogspot.com/-rNMvvrSuczg/UtbP8F695MI/AAAAAAAAFMM/7HSwCHokIzY/s1600/BCM+URI+Map.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="400" src="https://2.bp.blogspot.com/-rNMvvrSuczg/UtbP8F695MI/AAAAAAAAFMM/7HSwCHokIzY/s400/BCM+URI+Map.png" width="361" /></a></div><div class="separator" style="clear: both; text-align: left;"><br /></div>Next you need to configure your routing table. For any BCM or CS1000 endpoint you need to enter the destination digits and direct it to the NRS IP. I also selected MCDN Protocol CSE. Ensure that the port number matches up with what the CS1000 is expecting. For each IP Office configure the destination digits per your dialing plan and use the IP address of the IP Office.<br /><br />And now on to the fun piece. Configuring the IP Office is the most time consuming part of the whole setup. You need to configure three items: SIP Lines, Incoming Call Routes, and Short Codes.<br />For your SIP Lines you need a connection to each networked system. For simplicity I chose to use the same incoming call route, however the Line Group ID needs to be different for each site. In my example I used SIP Lines 17-20. All my Incoming Call Groups were 17, however I matched up the Line Group ID with the line number. Since these are IP trunks you can configure them however you like as long as they don't interfere with any other lines in the system. For the SIP line to the CS1000 use the Node IP address. For the BCM SIP Lines use the IP address of each individual BCM. Configure each SIP line with the maximum number of calls equal to the maximum number of SIP Line licenses in the system. This way each site is capable of having the maximum number of VoIP calls, as long as no other licenses are in use.<br />If you used the same incoming call route for each SIP trunk you only need to create a single incoming call route for IP calls. For the destination use a period (.) to have the call sent to whichever digits are being sent from the far end.<br />In order to make outgoing calls work you need to create short codes on the system to match the dialing plan. In my case we had four-digit dialing with each site having a unique first digit. This made it easy, I configured a short code for each first digit (i.e. 1XXX, 2XXX, 3XXX, etc.). Each short code was configured as Dial 3K1 using the appropriate line group for the far end site. The BCMs will not understand the standard dial string sent by the IP Office so you need to configure the phone context in the telephone number. In my example I showed the Private/CDP URI as cdp.udp so I needed the following Telephone Number in my short code:<br /><div style="text-align: center;"><span style="font-family: &quot;courier new&quot; , &quot;courier&quot; , monospace;">.";phone-context=cdp.udp"</span></div><div style="text-align: center;"><span style="font-family: &quot;courier new&quot; , &quot;courier&quot; , monospace;"><br /></span></div><div style="text-align: left;"><span style="font-family: inherit;">Once all of this is done just go ahead and commit your changes to the IP Office. It will probably require a reboot as you are changing IP information. Once everything comes up go ahead and make your test calls and give the customer a big smile on your way out the door, you're their hero.</span><br /><span style="font-family: inherit;"><br /></span><span style="font-family: inherit;">As always feel free to comment if you have any questions. I'll always do my best to help!</span></div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com2tag:blogger.com,1999:blog-8009054506756542917.post-37841975916636854672013-10-22T14:46:00.002-04:002013-10-22T15:13:00.000-04:00IP Office Initial StartupWhen powering up an IP Office for the first time a number of things are configured automatically. It is important to remember to power up all expansion cabinets first as the IP Office will reach out to all connected modules to identify them. Without power these devices can't be identified and the IP Office will not configure them.<br /><br />The IP Office will automatically build extensions and users for each extension port. This does not require the phones to be connected, simply that the port is installed and has power. The numbering starts at 201 and goes up sequentially, going from left to right across each port on the main unit and then sequentially up through the expansion modules. A default hunt group is created called MAIN with the number 200. The first ten users are added to this hunt group. By default all voice calls are directed to this hunt group. Data calls are routed to the RAS Access DialIn.<br /><br />All lines are assigned to Line Group 0 by default. A Short code of 9 is created to provide access to this line group.<br /><br />Embedded voicemail is configured by default. Every user will be assigned a mailbox and they will be active by default.<br /><br />The IP Office will configure IP Networking. LAN 1 will have the IP Address of 192.168.42.1 and LAN 2 will be 192.168.43.1. Both networks will have the Subnet Mask of 255.255.255.0 and will have the DHCP server configured with a 200 address range. When connecting to the network the IP Office will check to see if there is another DHCP server on the network. If another DHCP server is active the IP Office will disable the DHCP server for that interface.<br /><br />The IP Office System Name is configured using the MAC address of the IP Office control unit.Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-19054082302271861922013-10-01T20:54:00.000-04:002013-10-01T20:54:06.220-04:0065 key things to remember for the Avaya SBC Exam<br /><ol><li>Remote worker cluster using a Real Server IP and Real Server Port will use HTTP and HTTPS for registration requests.</li><li>Domain Policy Administration configures Rules, Policies, and Endpoit Policy Groups.</li><li>Trunk Server and Call Server Server Flows are administered in SIP Trunking.</li><li>Trunk Server and Call Server media interfaces are administered in SIP Trunking.</li><li>When integrated with IP Office the SSL VPN connection for SBC remote access is always on unless the connection to the Avaya VPN Gateway is down.</li><li>IP address and transport protocols are configured in a Server Profile.</li><li>Remote Worker is typically intended for remote users, travelling employees, and remote offices.</li><li>Before configuring SIP trunking the SBC must be installed and commissioned.</li><li>Topology Hiding Profiles hide source and destination IP addresses.</li><li>The Received Interface (as defined in Call Server Flow Administration) is an internal interface that receives inbound traffic.</li><li>Server Profiles define the connection and transport parameters from the SBC to the service provider and to the call.</li><li>Two routing profiles are required for a SIP trunk.</li><li>Before implementing SBC Remote Worker the SBCE&nbsp;+ Advanced Feature Set must be installed and commissioned.</li><li>Cluster Administration groups endpoints and applies similar attributes to them.</li><li>The SBC routing profile defines the next hop for SIP traffic.</li><li>A Server Profile applies to the trunk server and the call server.</li><li>The Supervisor account has the least system access.</li><li>To test the SBC configuration you should make internal-to-external and external-to-internal test calls.</li><li>When using an SBC with CM or CS1000 you should use SAL for remote access.</li><li>A SIP trunk requires a SIP Service Provider (ISTP), an SBC, and a Call Server.</li><li>With regards to application of Time of Day policies the packet is examined first, then the time of day is evaluated, after which the flow is determined and finally the policy is applied.</li><li>Best practice for SBC Remote Worker configuration is to fully encrypt both media and signalling.</li><li>Topology Hiding can be used to change SIP message parameters.</li><li>Media Interfaces define the IP addresses and ports for media.</li><li>To administer a signalling interface you must first add the signalling interface, then assign IP addresses and ports.</li><li>Session Manager can function as the Personal Profile Manager for a Remote Worker cluster.</li><li>Domain Policy Administration is where you would implement time of day restrictions.</li><li>Trunk Server and Call Server Signalling Interfaces are administered in SIP Trunking.</li><li>The Admin user has the highest privilege level to administer the SBC.</li><li>To verify a standalone installation you can log in to the SBCE GUI or log in the the SBC CLI.</li><li>Four networks and subnets are required to commission the SBC and SIP Trunking on an SBC located in a DMZ.</li><li>The Standalone SBC Software is the same as the SBC+EMS software.</li><li>A Standalone SBC requires the EMS+UC-Sec installation type.</li><li>HA Configurations always require a standalone EMS.</li><li>A Network Assessment is the only way to highlight a problem with inadequate bandwidth on the customer's network.</li><li>All SBC and EMS elements require Network Passphrase Authentication in a High Availability configuration.</li><li>HA installations are done using the installation type UC-Sec.</li><li>The EMS should be configured on M1 in a co-res (standalone) configuration.</li><li>A Site Survey will provide information on customer readiness.</li><li>In the EMS GUI you can identify the standby SBC by the SBC name.</li><li>The SBC is shipped with a serial cable or an RJ-45 to Serial cable to connect to the console port on the SBC for initial software installation.</li><li>In an HA SBC the M2 interfaces carry heartbeat information.</li><li>The SBC uses SSH port 222 for CLI access (Not port 22).</li><li>An SBC alarm will be triggered by System Level failures.</li><li>The SBC has a utility called Call Trace to monitor a single telephone.</li><li>The SBC Call Trace shows signalling level information for a specific URI (phone).</li><li>SBC incidents indicate a problem with the SBC software.</li><li>SBC Call Trace information can be viewed in the SBC GUI under Diagnostics\Protocol Tab.</li><li>A manually created backup can be called snapshot.zip.</li><li>You can look at System Incidents to find details of SBC operational problems.</li><li>SBC Alarm Descriptions can be found in the &nbsp;Appendices of Administration Guide of the SBC documentation.</li><li>The Link status/activity LEDs, hard drive LEDs, and over-temperature LEDs can give you a visual alert of a problem with the SBC.</li><li>SBC backups can be stored both locally and remotely.</li><li>You can view SBC System Alarms in the SBC GUI.</li><li>An SBC will show up as Ophaned in the EMS if there is a mismatch between the SBC and EMS software levels.</li><li>When upgrading an HA system you first upgrade the EMS, then the backup SBC, then the primary SBC.</li><li>Denial of Service (DoS) is an availability attack.</li><li>The DMZ is a neutral demarcation point between the trusted and untrusted network in which the SBC can be deployed.</li><li>A High Availability SBC will require a standalone SBC.</li><li>The SBC can be deployed inside the enterprise core or in a DMZ.</li><li>Advanced Services adds the Mobile Workspace (AKA Remote Worker) functionality.</li><li>Eavesdropping is an example of a confidentiality attack.</li><li>An SBC provides secure enterprise communications.</li><li>Signalling Manipulation is using the SIP Trunk Integration Module scripting language.</li><li>Signalling manipulation can modify the domain part of the SIP packet.</li></ol><div><br /></div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com3tag:blogger.com,1999:blog-8009054506756542917.post-37719917382468418662013-06-25T22:54:00.000-04:002013-06-25T22:54:47.652-04:00IP Office 8 Licensing<div class="MsoNormal">IP Office licensing is rather complicated. I’m not going to spend too much time on it but I would like to point out a few rather important details.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">The IP Office Basic Edition is a starter-edition of IP Office that only supports one PRI (Up to 64 total trunks with analog and SIP) and up to 9 Auto Attendants. It does not support any applications or IP telephones.&nbsp; It can be configured through a web interface or phone-based administration. Up to 100 telephones are supported on IP Office Basic Edition. Six concurrent connections to embedded voicemail are supported with approximately 25 hours of storage.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">IP Office Essential Edition offers more flexibility over the Basic Edition. It supports up to 40 Auto Attendants and 8 PRIs. This version offers the benefit of IP telephony, including One-X Communicator, mobile twinning, and remote worker support. The Essential Edition comes with 2 concurrent voicemail connections and 15 hours of storage. This can be upgraded to 4/20 or 6/25. There is a built-in conference bridge that supports up to 128 simultaneous connections. Up to 64 connections are supported in a single conference call.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">An IP Office Preferred license requires the Essential license and adds some additional functionality. Instead of the embedded voicemail Preferred Edition allows the use of Voicemail Pro. This can be on a standalone server or on a Unified Communications Module (UCM). Voicemail Pro adds significant functionality to the IP Office, including unlimited multi-level auto attendants and up to 40 simultaneous VM connections. The amount of storage available for VM Pro depends on the size of the hard drive on the VM server. A good benchmark is that each minute of voice stored will take up about one MB of hard drive space. VM Pro also includes call recording and an enhanced conference bridge. A single IP Office Preferred License can supply VM Pro to multiple sites in a Small Community Network configuration. VM Pro also adds support for web-based voicemail retrieval. The VM Pro server can be installed on a Linux-based or Windows-based server. The UC Module is Linux-based. When installing on a Windows server all Microsoft licensing requirements must be met for the server platform.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">IP Office Advanced Edition provides all of the features of the Preferred license and adds a number of contact center features. Advanced Edition supports up to 150 agents and 30 supervisors in a contact center environment. The Advanced Edition can provide access to real-time and historical reporting to manage efficiency. Recording of all calls can be configured with sufficient storage on your server.&nbsp; You can configure your menus to interact with an SQL database to provide a better interactive customer experience or to offer self-serve options. Visual Basic scripting is supported with a Windows-based Advanced Edition server.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">In addition to platform licensing you must also have a license for all your IP endpoints and trunks.&nbsp; A basic user (Digital or analog station) does not require a license. Any IP set requires an IP Endpoint License. There are two types of IP endpoint licenses: Avaya or third party. The appropriate license must be purchased for each endpoint. Additional licenses must be purchased for PRI B-channels above the eight that are included with the PRI daughterboard. <o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Currently license files are provided through <a href="http://adi.avaya.com/">http://adi.avaya.com/</a> but this will eventually be transitioned over to <a href="https://plds.avaya.com/">https://plds.avaya.com/</a><o:p></o:p></div><br /><div class="MsoNormal"><br /></div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-13697434076662452962013-06-25T22:04:00.002-04:002013-06-25T22:06:17.322-04:00IP Office Startup and default passwords<div class="MsoNormal">When the IP Office is powered on it will look for any attached hardware. This means that any attached hardware should be powered on prior to connecting power to the IP Office. When started up for the first time the IP Office will automatically build extensions and users for any recognized extension port, starting with extension/user 201. The IP Office will number extension from left to right on the IP Office then left to right on any attached modules, starting at Module 1 and working up. A hunt group (number 200) will be created with the first ten users as members. All detected lines are included in Line Group 0 and a short code on 9 is created to provide access to the default routing table. Embedded voicemail is also configured on startup. Every user on the IP Office receives a mailbox. This is also true when Voicemail Pro is enabled. <o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">The system name for the IP Office will be the MAC Address of the LAN1 port. As I previously mentioned the default IP addresses are 192.168.42.1 and 192.168.43.1 for LAN1 and LAN2, with a netmask of 255.255.255.0. A DHCP Server is built into the IP Office that is automatically configured to assign up to 200 IP addresses. The range is 192.168.42.2-201 and 192.168.43.2-201 for LAN1 and LAN2 respectively. If the IP Office detects that there is another DHCP server on the network it will disable the internal DHCP server. <o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">The IP Office also builds a few default usernames and passwords. They are as follows:<o:p></o:p></div><div class="MsoNormal">IP Office Administration: Administrator / Administrator<o:p></o:p></div><div class="MsoNormal">IP Office Security Settings: security / securitypwd<o:p></o:p></div><div class="MsoNormal">Remote Access Dialin: RemoteManager / password<o:p></o:p></div><div class="MsoNormal">System Password (for upgrades): password (no username)<o:p></o:p></div><div class="MsoNormal"><br /></div><br /><div class="MsoNormal">A number of other defaults are created as well. These include usernames that are used by the system for various functions as well as a few different levels of administration access. You can see the full list of users in the Security Settings. For obvious reasons you should consider changing some of these passwords. Do not change any of the passwords for system-level users (EnhTcpaService, SCN_Admin, IPDECTService, and SMGRB5800Admin). To change these passwords you need to open Manager and click File -&gt; Advanced -&gt; Security Settings. Log in using the Security Settings information. You can change the default passwords and create new users with various permissions from this screen.&nbsp; In the event that you do not have any passwords for your IP Office you can use a physical connection to the RS-232 port on the back of the IP Office control unit. You can connect to the RS-232 port by configuring a terminal to connect at 38,400/8/N/1, Flow Control Off, TTY or VT100. The command type to use is at, followed by the type at-securityresetall. The IP Office will prompt for a complex response after which all passwords will be defaulted.<o:p></o:p></div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-84763483714272969942013-06-25T21:22:00.002-04:002013-06-25T21:23:12.067-04:00IP Office Telephones<div class="MsoNormal">The IP Office supports analog, digital, and IP sets. Avaya uses the second digit of the phone model to indicate the phone type. A phone with the numbering scheme x4xx or x5xx is a digital set and a numbering scheme of x6xx is an IP set. Avaya uses the same sets for the IP Office as they do for Avaya Call Manager.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Avaya offers expansion modules known as Button Modules for some of their digital and IP sets. <o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">1400 series sets are digital sets using the Avaya Digital Control Protocol. Sets in this series include the 1403, 1408, and 1416 sets. &nbsp;The 1416 set supports up to three DBM32 expansion modules with external power. The 1400 series sets have a red light that shows up next to the first line appearance on the phone. These sets do not have self-labeling buttons and require a paper label for the buttons. Labels can be printed from within IP Office Manager using the DESI software available for free download from <a href="http://www.desi.com/">http://www.desi.com/</a><o:p></o:p></div><div class="MsoNormal">9500 series sets are digital sets that have soft-labels. The button labels that are programmed into the IP Office are passed through using the Digital Control Protocol to the phones. Available sets are the 9504 and 9508.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">1600 Series sets use the H.323 protocol and are visually similar to the 1400 series sets. They also require the use of DESI labels for button labeling. In addition to the 1600 sets there are 1600i sets that have a larger screen that support non-English lettering.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">9600 series sets include the 9620, 9630, 9640, and 9650 IP telephones. There are letters after the model number to indicate functionality. C indicates a colour display, L indicates low power consumption (PoE Class 1), G indicates Gigabit Ethernet. These sets support the SBM32 expansion module and require Professional Edition licensing. 9608, 9611, 9621, and 9641 sets are newer IP phones that are also supported. The 9600 series sets have displays that support soft labels so no DESI strips are required. <o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">The IP Office supports Nortel (now Avaya) 1120 and 1140 sets, as well as the newer 1220 and 1240 sets. <o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Avaya has three conference phone offerings. The B149 is an analog conference phone. The B159 is an analog conference phone with a USB connector for attaching to a PC or cell phone. The B179 is a SIP-enabled PoE conference phone that supports the G.722 Codec for HD voice quality. The B179 requires an Avaya IP Endpoint license to function and is supported on IP Office 7.0 and higher.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Avaya supports DECT R4 sets with the IP Office. Each base station can support up to eight simultaneous calls. DECT sets use the x7xx numbering scheme.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Avaya Video phones (1010, 1020, 1030, 1040, and 1050) are available. These video phones support internal video calls only (no, your customers won’t be able to see you). The Avaya softphone also supports video calls. The softphone can be installed on a Windows or MAC computer. The installer is located on the Admin CD.<o:p></o:p></div><div class="MsoNormal"><br /></div><br /><div class="MsoNormal">Legacy IP Office phones include the 4600, 5600, 4400, and 5400 series sets that are also supported. These sets are not available for new sales.&nbsp;<sub><o:p></o:p></sub></div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-24456873104937714782013-06-25T19:20:00.002-04:002013-06-25T19:38:38.788-04:00IP Office HardwareThe IP Office 500 v2 is the base system for all new IP Office installations. It supports up to 384 extensions (ports) and up to 1000 subscribers (users).<br /><div class="MsoNormal"><o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">The IPO500v2 has a LAN and WAN port, however in the IPO programming these are referred to as LAN1 and LAN2 respectively. The default IP addresses for these ports are 192.168.42.1 and 192.168.43.1 respectively, with a netmask of 255.255.255.0.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Up to four base cards can be installed in the system. Each base card can be fitted with a trunk daughterboard, with the exception of the 4-port expansion module. <o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">A Digital Station card supports up to&nbsp;&nbsp; eight digital stations. Up to three DS card can be included in the IP Office.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">A TCM8DS Module supports up to eight Nortel 7000-series digital stations on the IP Office. <o:p></o:p></div><div class="MsoNormal">Voice Compression Module (VCM) provides resources for use with IP transcoding (same as DSP resources in the Nortel world). There are two VCM base cards – 32- or 64-VCM cards.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">A Combo Card supports six digital stations, 2 analog stations, four analog trunks, and 10 VCMs. You can have up to two combo cards in an IP Office. Port 8 on a combo card can also be configured as a powerfail line.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">A P-2 or P-8 base card supports 2 or 8 phone sets. The P2 is often used if a trunk module is required without any other functionality as it is the least expensive card available for the IP Office.<o:p></o:p></div><div class="MsoNormal">The 4-port expansion module allows for additional expansion modules to be connected. It must be plugged into Slot 4 of the IP Office.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">The Unified Communications Module is an embedded Linux server that can host Voicemail Pro (up to 200 users and 40 ports) and the One-X Portal. It is managed separately from the IP Office through a web interface on Port 7070. The UCM is not supported with IP Office Basic Edition.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">The IP Office supports up to twelve expansion modules. The IPO500v2 has eight ports built into the back for expansion and an optional 4-port expansion module can be installed into slot 4 of the IP Office to add the extra four expansion modules.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Expansion Modules for the IP Office include 16-Port Analog Trunk, BRI So8, 16- or 30-port Analog Station, 16- or 30-port Digital Station, and 16- or 30-port Amphenol Digital Station for Nortel 7000 series phones.<o:p></o:p></div><div class="MsoNormal"><br /></div><div class="MsoNormal">Daughterboards are connected to the IPO Base Cards to allow for trunking. The possible options are 4-port analog trunk, BRI (not usually used in North America), or PRI (1- or 2-port PRI). By default the IPO PRI module only comes licensed for eight B-Channels. Additional licenses will be required to open channels 9-23.<br /><br />The IP Office has two slots for SD Cards in the back of the unit. The System SD card contains a feature key that is tied to licensing. The IP Office will run for up to two hours with the System SD Card removed in unlicensed mode. IP Office voicemail is hosted on the SD card so voicemail will not function while the card is removed. A secondary SD Card can be used to back up data and host voicemail. The secondary slot supports SDHC Cards 4GB and greater.&nbsp;</div>Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-80584886975304289612013-03-11T21:25:00.000-04:002013-03-11T21:25:02.639-04:00Changing the time and date on an IP Office<br />So Daylight Savings Time has begun in most of North America. As a result I have seen a few inquiries about changing the time on the IP Office.<br /><br />The IP Office supports automatic daylight savings time changes.The dates are configured in IP Office Manager:<br /><br />Under System, scroll down to see the Time Settings box<br />Ensure that Automatic DST is checked<br />Check the dates for 2013 (you may need to use the drop down box to see the correct year)<br />If required hit the Edit button to change the dates.<br /><br />If you do not have Voicemail Pro the time source can be set manually. Voicemail Pro configuration will match the system time to the time on the Voicemail Pro PC. Ensure that the time zone matches the Voicemail Pro server. If you have a PC that is always running Manager you can use this as a time source as well. The IP Office will retrieve the time from that source.<br /><br />SNTP can be configured as another time source. This will retrieve the time from a PC on your network or from an Internet Time Server if your IP Office is connected to the Internet. If anyone requires assistance setting this up just say something in the comments and I'll do a post all about SNTP.<br /><br />If the time on the IP Office is off by too much the time will not sync properly, regardless of your time source. To correct this you will have to change the time source to None, configure the time, then return the IP Office to your chosen time source.<br /><br />Now on to the good stuff. How to change the time on an IP Office system.<br /><br /><br /><div class="MsoListParagraph" style="mso-list: l0 level1 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->1)<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Check the time source on the IP Office. To change the time manually the time source must be set to None.<o:p></o:p></div><div class="MsoListParagraph" style="mso-list: l0 level1 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->2)<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Ensure that the time zone and UTC Offset are correct.<o:p></o:p></div><div class="MsoListParagraph" style="mso-list: l0 level1 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->3)<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Select a user to have System Phone rights. This can be any 1400, 1600, 9500, or 9600 phone except for the 1403 and 1603 model.<o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->a.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->In IP Office Manager, select the user<o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->b.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Under <b>System Phone Rights</b>select <b>Level 2</b><o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->c.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Select the <b>Button Programming</b>tab<o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->d.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Add a button with the action of <b>Emulation </b><b><span style="font-family: Wingdings;">à</span></b><b> Self Administer</b>, and enter <b>2 </b>&nbsp;for the <b>Action Data</b> <o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->e.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Label the button appropriately, I usually label it Administer<o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->f.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Apply the changes and merge the configuration to the IP Office<o:p></o:p></div><div class="MsoListParagraph" style="mso-list: l0 level1 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->4)<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Go to your newly configured phone and press the Administer button that you programmed. The option to program the Time and Date will be displayed. <o:p></o:p></div><div class="MsoListParagraph" style="mso-list: l0 level1 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->5)<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Check the date by pressing <b>Date</b> or the time by pressing <b>Time</b>. <o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->a.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->The date is entered using the format specified in the system, using # or * to insert the / separators<o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->b.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->The time is entered using 24 hour format, using # or * to separate hours and minutes <o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->c.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->The key labeled &lt;&lt;&lt; can be used to backspace<o:p></o:p></div><div class="MsoListParagraph" style="margin-left: 1.0in; mso-list: l0 level2 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->d.<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->Press Done when finished<o:p></o:p></div><div class="MsoListParagraph" style="mso-list: l0 level1 lfo1; text-indent: -.25in;"><!--[if !supportLists]-->6)<span style="font-size: 7pt;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><!--[endif]-->The phone will return to idle when you press done. You can remove the Administer button at this point if you want to prevent an accidental configuration change.<span style="color: #1f497d;"><o:p></o:p></span></div><br /><br />Once again I hope this helps! Comments and questions are always welcome.<br /><br />Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com2tag:blogger.com,1999:blog-8009054506756542917.post-48374082937775036482013-02-21T20:06:00.000-05:002018-04-30T12:59:33.449-04:00IP Office Music on HoldI've seen a lot of questions regarding Music on Hold on the IP Office. It's not nearly as simple as music was on the BCM but it's still quite easy. Starting with IP Office 3.1 you can use an external source There's always the easiest solution - plug it in to an audio source. There is a 3.5mm jack on the IP Office Control Unit. This will allow you to play anything and is a frequent solution for people who are happy to have the radio play as their hold music. There is no real effort required here - you just plug it in and point the IP Office to use this source and PRESTO! you have hold music. You can also configure an internal analog trunk as a music source. I've never seen this implemented myself, however my experience is quite limited. I'm guessing this would be an easy way to integrate a third party MoH source. This is supported on IP Office 6.1 and up. The way that seems to confuse people is using the internal source. This is a WAV file that is housed on the IP Office. There are some rather severe limitations here. The IP Office 500 and 500 V2 can support up to 90 seconds of recording time. The IP401 does not support an internal music source. Up to Release 4.2 there was only one supported source. In release 4.2 and up there are up to four supported sources, This is also the release in 90 second length is supported. Prior to Release 4.2 you can only have up to 30 seconds. Now on to loading the hold music. The IP500 V2 system can have the hold music loaded during the initial setup. I'm copying this right out of the Avaya documentation: <br /><blockquote>By default the IP Office will use internal music on hold by uploading a music file from the IP Office Manager PC. For IP500v2 systems, you can load a file onto the System SD card prior to installing it in the IP Office. The file must be of the following format and must be called holdmusic.wav. 1. Rename the music file holdmusic.wav. 2. Using a card reader, copy the file into the /system/primary folder on the System SD memory card. 3. If the IP Office is or will be configured for additional hold music files (up to 3 additional files), copy those files to the same location. The name of the additional files must match those specified in the IP Office system's configuration. </blockquote>Post-install the Music on Hold can be added using Embedded File Management. Just switch to Embedded File Management mode and copy the file to the IP Office SD Card and upload holdmusic.wav to the PRIMARY folder on the SD Card. The other method of loading the IP Office hold music is to use the TFTP Server. When the IP Office is rebooted it will always look for a copy of holdmusic.wav on a TFTP server (if present). The IP Office Manager acts as a TFTP Server while running so this is an easy solution that requires no additional software. The file can just be placed in the working directory of the IP Office Manager software. If you need assistance in configuring additional MoH source files (IPO 4.2+) it is in the documentation. I hope this little bit helps people with their questions. <br /><br />The IP Office supports multiple hold music selections, based on the incoming call route. You can use a WAV file or an analog extension to provide the music. In order to use different WAV files you need to add them to the control unit using embedded file management. You can enter a friendly name for the source. The source number is specified by the system, starting with 2.<br /><br />Specify a WAV file using WAV:filename.wav. Specify an extension using XTN:222 (replace 222 with the extension number). When using an extension you need to change the extension to show that it is a music on hold source.<br /><br /><div class="separator" style="clear: both; text-align: center;"><a href="http://3.bp.blogspot.com/-1qzblMN_-u8/VLgdTUSjo_I/AAAAAAAAQZQ/yWmdhcuVmF0/s1600/HoldMusic.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="352" src="https://3.bp.blogspot.com/-1qzblMN_-u8/VLgdTUSjo_I/AAAAAAAAQZQ/yWmdhcuVmF0/s1600/HoldMusic.png" width="640" /></a></div><div class="separator" style="clear: both; text-align: center;"></div><br />Once you have added your sources you can add them to the incoming call route:<br /><br /><div class="separator" style="clear: both; text-align: center;"><a href="http://3.bp.blogspot.com/-ZoYkA9Oq8YY/VLgdseQeVqI/AAAAAAAAQZY/JokXLfg3OJA/s1600/ICR-holdmusic.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="458" src="https://3.bp.blogspot.com/-ZoYkA9Oq8YY/VLgdseQeVqI/AAAAAAAAQZY/JokXLfg3OJA/s1600/ICR-holdmusic.png" width="640" /></a></div><br />Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-76472871308270118462013-01-17T23:49:00.000-05:002013-01-17T23:49:14.655-05:00BCM and IP OfficeI've moved to a new city and started a new job. They hired me based on my experience with the CS1000, but my primary role is working with small systems like the BCM. The BCM is a lot easier to manage than the CS1000 ever could be, but you sacrifice the ability to configure everything. That being said, it's a great small system. Avaya has ended new sales of this hardware platform, instead pointing people towards the IP Office for Small / Medium Business markets. The IP Office adds a lot more configuration options, and with the Voicemail Pro add-on it can be configured to function with incredible call routing and detail. The BCM is managed by a program called Business Element Manager. It's a standalone application that can be downloaded by navigating to the IP address of the BCM. Once downloaded it installs quickly. You can manage a large number of BCM systems from a single Business Element Manager installation, but if you need to manage many sites I would suggest you use the option to create folders for each grouping, thereby making management easier. I'll spend more time on the BCM at a later date. I'll also put some effort into the IP Office, as this is the future of Avaya's SMB market.Aaron Dyckhttps://plus.google.com/109378934310868253840noreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-54911189205331274412009-08-20T15:16:00.000-04:002009-08-20T15:17:48.784-04:00The Layer 2 SwitchThe final component that is required to set up a functional CS1000E System is the Layer 2 switch. This provides data connectivity between the components of the system, including connectivity to IP sets.<br /><br />A layer 2 switch transmits data packets directly to the targeted device only, rather than broadcasting them to every device on the network (like a hub would do). This prevents collisions and packet loss on the data network. Nortel recommends the use of a Baystack 460 switch with Power over Ethernet (for installations that use PoE), or the Baystack 470-24T switch.<br /><br />Any third party layer 2 switch can be used with the CS1000 system, as long as it supports VLAN tagging and QoS. A switch that does not support QoS is not suitable for VoIP applications.<br /><br />For a Power over Ethernet scenario each IP device requires a device that will split the CAT5 cable to add a power connection. PoE is useful in creating a system that will function in the event of a power failure without the use of a UPS at each station.Nortel Nerdnoreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-25890873308993801572009-08-20T14:48:00.000-04:002009-08-20T14:49:55.453-04:00The Media Gateway 1000 ChassisThe Media Gateway 1000 Chassis houses the circuit cards and connectors that support both IP and TTDM functionality. Slot 0 must always contain the MGC, which acts as a gateway controller for IP Media Gateways (IPMG). The MG1000 Chassis has four additional universal slots that can house various other cards, including the CP PM Call Server and CP PM Signaling Server, as well as line and trunk cards. The MG1000E Chassis also supports the addition of a Media Gateway Expander, which provides an additional four slots (7-10). Slots 5 and 6 are not used. The MG1000E Chassis connects to the MGE via two copper cables, labeled CE-MUS and DS-30X. The MGE does not support digital trunks or analog clock controllers.<br /><br />The CS1000E system supports up to a maximum of 50 MG1000 and MGE pairs. The MG1000E can support the following cards:<br />Voice Gateway Media Cards – these transcode between the IP network and digital circuit cards<br />Service Cards – These provide services such as music on hold and recorded announcements<br />Analog Line Cards – support analog stations, fax machines, and modems<br />Digital Line Cards – support digital phones<br />Analog Trunk Cards – used to connect the system to an analog network, such as the PSTN<br />Digital PSTN Interface Cards – this can include E1, T1, and ISDN BRI cards to provide access to the PSTN<br />Class Modem Cards – such as the XCMC<br />DECT Mobility Cards – these provide wireless functionality<br /><br />It is important to remember that digital trunks, PRI and BRI ISDN cards, and DECT mobility cards are only supported in the MG1000E that contains the MGC. They can not be placed in the MGE.Nortel Nerdnoreply@blogger.com4tag:blogger.com,1999:blog-8009054506756542917.post-21163599329238015722009-08-20T14:40:00.000-04:002009-08-20T14:41:37.257-04:00Media CardsThere are two types of media cards that can be used to provide additional DSP resources to the MG1000E. The Media Card 32 (MC32) and the Media Card 32 Security (MC32S) both provide 32 channels of DSP Resources to the system. The difference between the two is that the MC32S provides the addition of Secure Real-Time Protocol (SRTP). SRTP encrypts the IP media path to and from the DSP channels on the MC32S card. It also offers improved echo performance over the MC32.Nortel Nerdnoreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-63541410954227659352009-08-20T14:37:00.001-04:002009-08-20T14:38:36.496-04:00Media Gateway ControllerThe Media Card Gateway controls the interface cards and application cards in the MG1000E Chassis. On a hardware level the MGC provides conference channels and tone generation for the phones. DSP Daughterboards are connected to the MGC to provide tone and conference functions between TDM and IP phones. The MGC is a replacement for the legacy Small Systems Controller (SSC). It has about 10 x the processing power of the SSC and four times the memory. It uses a standard Compact Flash for internal storage and has two internal ports for the installation of DSP Daughterboards. There are a total of six Ethernet ports on the MGC, four on the front (two ELAN, two TLAN), and two that are available through the backplane via a 50-pin to Serial/ELAN/TLAN Adapter.<br /><br />There are two types of DSP Daughterboards that can be placed into the MGC. The 96-port DSP Daughterboard can be placed in Slot 2 only, and provides 96 channels of voice. The 32-port DSP Daughterboard can be placed in either Slot 1 or Slot 2. Using one or two DSP32 DBs you can achieve 32 or 64 channels of voice, and using a standalone DSP96 DB or a DSP32 and a DSP96 you can achieve 96 or 128 channels of voice. If the system requires more than 128 channels of voice to be converted from IP to TDM a separate Voice Gateway Media Controller card will be required. The DSP resources provide VoIP CODECs, compression, and echo cancellation.<br /><br />The MGC provides 60 channels of Tone and Digit Switching (TDS), sixteen Digitone Receivers (DTR) or Extended Tone Detector (XTD) units, and additional tone service ports. It also provides two loops of 30 conference units each, for a total of 60 conference channels.Nortel Nerdnoreply@blogger.com0tag:blogger.com,1999:blog-8009054506756542917.post-87587867898493200962009-08-20T13:50:00.000-04:002009-08-20T13:51:40.458-04:00The Signaling ServerA CS1000E system can have one of four types of signaling servers. There are two Signaling Servers that are available from a third party and there is the CP PM Signaling Server. In addition, there is support for the legacy ISP1100 Signaling Server.<br /><br />The two COTS Signaling Servers that can be used are the IBM X306m Signaling Server and the HP DL320-G4 Signaling Server. Both of these servers can be configured using either Linux or the VxWorks software that the Nortel equipment uses. If the COTS signaling server is using Linux it adds the functionality of a tool called Enterprise Common Manager that allows for more advanced configuration options through a web interface than the default Nortel software allows through Element Manager.<br />The ISP1100 Signaling Server was used with previous releases and is supported for backwards compatibility. It is now discontinued and is not recommended, although it will function as a signaling server.<br /><br />The CP PM Signaling Server uses the same hardware as the CP PM Call Server. There are a few small differences, however. The most noticeable difference is that the Signaling Server has a 40 GB Hard Disk Drive and no on-board Compact Flash. To enable the system to use the 40 GB Hard Disk Drive to load software it is necessary to set Switch 5 to position 2. Additionally, there is no security dongle on the Signaling Server.<br /><br />The CP PM Signaling Server is connected either directly to the MGC or to a layer 2 switch using the ELAN and TLAN ports. Nortel recommends connecting them via a layer 2 switch. The ELAN port is used for address, data, and control signals between the various components of the CS1000E system. The TLAN port is used for telephony signaling traffic. The HSP port is not used on a signaling server. The indicator lights are the same as those on the CP PM Call Server.<br /><br />The role of the signaling server is to provide SIP and H.323 signaling, IP phone signaling, and IP peer networking. The signaling server can use either VxWorks of Linux to perform these functions. The CP PM Signaling Server can only use VxWorks. A signaling server also provides the H.323 Gateway software to provide IP Peer Networking features between multiple MG1000Es and the CS1000E Call Server. In addition, it provides virtual trunk application and support for IP terminals. It can be configured with a survivability configuration and a load-sharing configuration. Each signaling server can support up to 5,000 IP users and a maximum of 1,800 SIP and H.323 trunks. Of these 1,800 trunks, a maximum of 1,200 H.323 trunks are available. This maximum can only be achieved if H.245 tunneling is enabled. If the server is running Personal Directory, Redial List, and Caller List, a maximum of 1,000 IP users are supported per Signaling Server.<br /><br />A Signaling Server has a number of software applications that are installed to provide functionality. This includes the SIP/H.323 Gateway Signaling Software (for virtual trunks), IP Line Software (including the Terminal Proxy Server), a Network Routing Service (NRS) application, the Element Manager web server, the NRS Manager web server, and an application server that includes Personal Directory, Callers List, and Redial List.Nortel Nerdnoreply@blogger.com0